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Supports: 3GP, 3G2
3GP is the container 3G-era handsets recorded to, standardized by 3GPP on top of the ISO base media file format — the same family as MP4. Its audio track is almost always AMR speech or AAC, neither of which most audio editors open directly. This tool decodes that track and writes it out as WAV: uncompressed linear PCM in a RIFF wrapper, which every DAW, transcription tool, and sampler reads without a codec. What it cannot do is add fidelity — a narrowband voice memo stays narrowband, just in an editable, universally supported container.
| Source codec | Sampling rate | Audio bandwidth | Codec bitrate | Typical origin | What the WAV gives you |
|---|---|---|---|---|---|
| AMR-NB (narrowband) | 8 kHz | roughly 200–3,400 Hz | 4.75–12.2 kbit/s | Voice memos, recorded calls | An exact PCM copy of speech-band audio |
| AMR-WB (ITU-T G.722.2) | 16 kHz | roughly 50–7,000 Hz | 6.6–23.85 kbit/s | HD-voice recordings | Fuller speech, still no high treble |
| AAC-LC | up to 48 kHz | full range | typically 64–256 kbit/s | Camcorder and video-clip audio | Music-grade PCM, editor-ready |
| HE-AAC v1 / v2 | up to 48 kHz | full range, band-replicated above ~10 kHz | typically 24–64 kbit/s | Low-bitrate streaming clips | Full-range PCM, but reconstructed highs |
| Property | Value |
|---|---|
| Container | RIFF (Resource Interchange File Format), chunk-based |
| Published by | IBM and Microsoft, 1991 |
| Payload we write by default | Linear PCM, 16-bit little-endian |
| Byte order | Little-endian (AIFF is the big-endian sibling) |
| Compression | None — size is fixed arithmetic, not content-dependent |
| Maximum file size | Under 4 GiB, because the RIFF header stores size in a 32-bit unsigned integer (about 6.8 hours of CD-quality stereo) |
| Metadata | Optional INFO chunk, XMP, or ID3 in extra chunks; many apps ignore them |
| Best for | Editing, transcription, sampler and DAW import, mastering input |
.3gp or .3g2 clip onto the page, or click "+ Add Files" to browse. You can queue several at once and convert them with the same settings.Because WAV is uncompressed, its size is pure arithmetic: sample rate multiplied by bit depth multiplied by channel count. That makes the two settings above the only real levers on output size, and both are lossless-neutral in the direction that matters.
Because the source is almost certainly AMR-NB, the narrowband speech codec 3G phones used for calls and voice memos. AMR-NB samples at 8 kHz and carries roughly 200–3,400 Hz, which is the telephone speech band. Sibilants, high-pitched voices, and any music above that cutoff were discarded when the phone recorded. WAV stores the decoded samples exactly, so it preserves what is there but cannot reconstruct what was never captured.
Considerably, and you can calculate it exactly. Uncompressed PCM runs at sample rate x bit depth x channels: CD-quality stereo (44,100 Hz, 16-bit, 2 channels) is 1,411.2 kbit/s, or about 10.1 MB per minute. A mono 8 kHz 16-bit WAV is 128 kbit/s, about 0.96 MB per minute. An AMR-NB source at 12.2 kbit/s is roughly a tenth of even that mono figure, so expect the WAV to be many times the original.
Only when a downstream tool requires a standard rate. Upsampling interpolates the existing samples onto a denser grid; it does not reveal detail the 8 kHz recording never had, and it multiplies the file size by more than five. If your editor accepts arbitrary sample rates, ORIGINAL keeps the file smaller and avoids an unnecessary resample.
Lossless relative to the decode, not relative to the original sound. The phone's AMR or AAC encoder permanently discarded data at capture time, and that loss is baked in. From the decoded PCM forward, WAV adds nothing further: our converter writes 16-bit little-endian PCM that stores those samples exactly, with no second lossy stage.
16-bit linear PCM, little-endian, which is the default payload for WAV output here and the depth every editor and transcription tool accepts. That matches the audio-CD standard and is more than enough headroom for a source that was a low-bitrate speech or AAC stream. There is no benefit to a deeper output from an already-compressed phone recording.
Neither. WAV is an audio-only container, so the video track is dropped entirely — the audio is decoded and re-written on its own, and nothing of the picture survives. WAV can technically carry an INFO chunk or ID3 tags, but the title, artist, and date fields a 3GP might hold are not reliably mapped across, and many players ignore WAV metadata anyway.
That is a hard limit of the format, not of the conversion. WAV stores the file size in a 32-bit unsigned integer inside the RIFF header, so a WAV cannot exceed 4 GiB — roughly 6.8 hours of 44.1 kHz 16-bit stereo, or far longer for mono speech. For anything approaching that length, trim the clip into sections or use a compressed format instead.
WAV when you are going to edit, transcribe, or process the audio, because it avoids a second lossy generation. MP3 when you just need to store or send it — an AMR-NB voice memo is speech-band audio to begin with, so a modest MP3 bitrate loses almost nothing audible and the file stays small. Use 3GP to MP3 for that path, or compress the WAV afterward.
Use the AMR to WAV converter instead. Many Android phones and voice-recorder apps save the AMR speech stream without the 3GP wrapper, and that tool takes the raw .amr directly. The output and the fidelity expectations are identical to the ones described here, since the underlying speech codec is the same.
Files upload over an encrypted connection, are processed on our servers, and are deleted automatically after a few hours — no sign-up, no watermark, never shared or made public. The practical limit on a long recording is upload size and time rather than anything in the WAV format itself.